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// Copyright 2019 The Chromium Authors. All rights reserved.
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.
#ifndef THIRD_PARTY_BLINK_RENDERER_PLATFORM_WEBRTC_WEBRTC_SOURCE_H_
#define THIRD_PARTY_BLINK_RENDERER_PLATFORM_WEBRTC_WEBRTC_SOURCE_H_
#include "third_party/blink/renderer/platform/platform_export.h"
namespace base {
class UnguessableToken;
}
namespace media {
class AudioBus;
}
namespace blink {
class WebRtcAudioRenderer;
// TODO(xians): Move the following two interfaces to webrtc so that
// libjingle can own references to the renderer and capturer.
class PLATFORM_EXPORT WebRtcAudioRendererSource {
public:
// Callback to get the rendered data.
// |audio_bus| must have buffer size |sample_rate/100| and 1-2 channels.
virtual void RenderData(media::AudioBus* audio_bus,
int sample_rate,
int audio_delay_milliseconds,
base::TimeDelta* current_time) = 0;
// Callback to notify the client that the renderer is going away.
virtual void RemoveAudioRenderer(WebRtcAudioRenderer* renderer) = 0;
// Callback to notify the client that the audio renderer thread stopped.
// This function must be called only when that thread is actually stopped.
// Otherwise a race may occur.
virtual void AudioRendererThreadStopped() = 0;
// Callback to notify the client of the output device the renderer is using.
virtual void SetOutputDeviceForAec(const std::string& output_device_id) = 0;
// Returns the UnguessableToken used to connect this stream to an input stream
// for echo cancellation.
virtual base::UnguessableToken GetAudioProcessingId() const = 0;
protected:
virtual ~WebRtcAudioRendererSource() {}
};
// TODO(xians): Merge this interface with WebRtcAudioRendererSource.
// The reason why we could not do it today is that WebRtcAudioRendererSource
// gets the data by pulling, while the data is pushed into
// WebRtcPlayoutDataSource::Sink.
class PLATFORM_EXPORT WebRtcPlayoutDataSource {
public:
class Sink {
public:
// Callback to get the playout data.
// Called on the audio render thread.
// |audio_bus| must have buffer size |sample_rate/100| and 1-2 channels.
virtual void OnPlayoutData(media::AudioBus* audio_bus,
int sample_rate,
int audio_delay_milliseconds) = 0;
// Callback to notify the sink that the source has changed.
// Called on the main render thread.
virtual void OnPlayoutDataSourceChanged() = 0;
// Called to notify that the audio render thread has changed, and
// OnPlayoutData() will from now on be called on the new thread.
// Called on the new audio render thread.
virtual void OnRenderThreadChanged() = 0;
protected:
virtual ~Sink() {}
};
// Adds/Removes the sink of WebRtcAudioRendererSource to the ADM.
// These methods are used by the MediaStreamAudioProcesssor to get the
// rendered data for AEC.
virtual void AddPlayoutSink(Sink* sink) = 0;
virtual void RemovePlayoutSink(Sink* sink) = 0;
protected:
virtual ~WebRtcPlayoutDataSource() {}
};
} // namespace blink
#endif // THIRD_PARTY_BLINK_RENDERER_PLATFORM_WEBRTC_WEBRTC_SOURCE_H_
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